1. Field of the Invention
This invention relates to an acoustic echo canceller for use with communication lines, indoor sound field controllers, and high-quality speech communication conference systems, and for cancelling an acoustic echo component produced when a signal on a receiving communication line appears on a transmitting communication line via an acoustic echo path.
2. Description of the Related Art
Generally, the acoustic echo cancellers are roughly classified into those for cancelling an echo produced due to an impedance mismatch of a 2-wire 4-wire converter on long-distance telephone lines using a communication satellite and submarine cables and those for cancelling an echo produced due to acoustic coupling of speaker speech at a loudspeaking telephone set in a TV conference system, etc., each of which includes a correction amount calculation circuit, a variable coefficient filter for generating an artificial acoustic echo, and a subtractor. The basic operation of the acoustic echo canceller will be described hereinafter.
FIG. 1 shows the basic configuration of an acoustic echo canceller. A receive signal input terminal 1 is connected to a receive signal output terminal 2 and a receive signal at the receive signal input terminal 1 is branched to a variable coefficient digital filter 3 for generating an artificial echo. A transmit signal from a transmit signal input terminal 4 and the artificial acoustic echo which is an output of the variable coefficient digital filter 3 are input to a subtractor 5 for cancelling the acoustic echo component in the transmit signal. An output of the subtractor 5 is sent to a transmit signal output terminal 6. An output of the transmit signal output terminal 6 and the signal at the receive signal input terminal 1 are input to a correction amount calculation circuit 7. The filter coefficient of the variable coefficient digital filter 3 is corrected in response to an output of the correction amount calculation circuit 7. In the variable coefficient digital filter 3, the receive signal is input to a receive signal input register 8 and a sum-of-products operation on the receive signal in the receive signal input register 8 and an artificial impulse response in an artificial impulse response register 9 is performed by a sum-of-products operation circuit 10. The result of the sum-of-products operation circuit 10 is output as an artificial acoustic echo. The receive signal output terminal 2 and the transmit signal input terminal 4 are connected to a 2-wire 4-wire converter on a long-distance telephone line or connected to a loudspeaker and a microphone in a loudspeaking telephone system.
Assume that the signal propagation characteristic of an acoustic echo path can be represented as a linear form and by an FIR type digital filter. Let its impulse response be h(t), input receive signal be x(t), and sampling time interval be T. Acoustic echo at time kT, y.sub.k, is represented as follows: EQU Y.sub.k =h.sub.k 'x.sub.k ( 1)
where EQU h=[h.sub.1, h.sub.2, . . . , h.sub.n ]' EQU x=[x.sub.k-1, . . . , x.sub.k-n ]' (2)
': Inversion of vector
On the other hand, assuming that an estimated value of h at time kT is hs.sub.k, an estimated value of y.sub.k, ys.sub.k is given as follows: EQU ys.sub.k =hs.sub.k 'x.sub.k ( 3)
When a speech signal exists at the receive signal input terminal 1 and only an acoustic echo with no speech signal exists at the transmit signal input terminal 4, the acoustic echo canceller performs echo cancel operation as an adaptive operation state. Generally, a learning method for identification ("A Learning Method for System Identification" by Atuhiko NODA and Jin-ichi NAGUMO, Measurement and Control, Vol. 7, No. 9, pp. 597-605 (1968)) is adopted as an algorithm of the adaptive operation. Sequential correction of hs.sub.k by the learning method for identification is performed according to EQU hs.sub.k+1 =hs.sub.k +.alpha.(x.sub.k e.sub.k)/x.sub.k 'x.sub.k ( 4)
where EQU e.sub.k =y.sub.k -ys.sub.k, 0&lt;.alpha..ltoreq.1 (5)
e.sub.k is called a remaining acoustic echo. Such calculation operation is performed in the coefficient correction amount calculation circuit 7. A variable coefficient series hs.sub.k is stored in the artificial impulse response register 9. .alpha. is a correction loop gain for determining sensibility of estimation; the nearer to 1.0 the value, the greater given the correction amount, enabling an acoustic echo to be cancelled at a high speed. However, for actual use, the value must be changed depending on near-end noise and the line state. It is common practice to determine the correction loop gain .alpha. according to a rule of thumb at present.
When the acoustic echo characteristic in a loudspeaking sound field is represented by such FIR type digital filter, a large configuration of several hundreds to several thousands of taps results and the operation amount involved in updating the correction amount of the variable coefficient series hs.sub.k becomes enormous and cannot be covered by a small-scaled hardware. Thus, the variable coefficient series hs.sub.k is divided into several stages for processing and the operation amount for updating in one step is reduced (for example, Japanese Patent Unexamined Publication No. Sho. 63-246934). As an example, FIG. 2 shows the acoustic echo cancellation characteristic with an autoregressive signal as an input when 2-division processing is performed for the variable coefficient series divided into first and latter halves. For comparison, a case where no division processing is performed is also shown. In the figure, "ERLE" is short for echo return loss enhancement. Assuming that the total of variable coefficient series is N, the division contents become as follows:
hs1.sub.k : 0 to N/2
hs2.sub.k : N/2+1 to N
By applying the above-mentioned division range, from expression (4), update algorithm can be represented as EQU hs1.sub.k+1 =hs.sub.1k +.alpha.(x.sub.k e.sub.k)/x.sub.k 'x.sub.k ( 6) EQU hs2.sub.k+1 =hs.sub.2k +.alpha.(x.sub.k e.sub.k)/x.sub.k 'x.sub.k ( 7)
which is an adaptive algorithm for updating all variable coefficient series hs.sub.k at M of two or in two steps (where M is the number of steps for updating all the coefficient series). Therefore, the operation amount in one step can be reduced to a half; of course, if the division count N is increased, the operation amount can be reduced to 1/N accordingly.
If processing of updating the correction amount of variable coefficient series hs.sub.k is performed with division, the operation amount involved in the updating is reduced, but the variable coefficient not updated in one step causes an estimation error to occur on generation of an artificial acoustic echo; resultantly, the remaining echo increases and the acoustic echo cancellation characteristic is degraded. As shown in FIG. 2, as compared with processing in which all variable coefficients are updated at a time, the updating processing with division requires about double time until saturation, and the convergence speed lowers to a half. When the convergence speed lowers, the remaining echo at a large level exists on the line, causing the communication state to be degraded. Also, if a path fluctuation occurs on the echo path during talking, the follow-up characteristic to that state worsens and the acoustic echo cancellation characteristic changes rapidly, causing rasping remaining voice to occur, so that high-accuracy and high-quality acoustic echo cancellation cannot be performed.